Existing audio signal processing systems suffer from a variety of limitations. Some of the limitations are imposed by the transmission or storage medium or other technological deficiencies; other limitations are the result of environmental issues. Regardless of the reason, the result is the same: the listener receives a less than optimal listening experience.
For example, the amplification necessary to hear the quietest portions of an audio signal may result in maximum amplitudes that are undesirably loud. Conversely, amplitudes allowing loud portions of an audio signal to be heard at a comfortable level, may result in not being able to hear quiet portions of the signal. Enabling the entire signal to be comfortably heard at all times by the listener requires that the input source dynamic range of the signal be transformed into the dynamic range of the listener's environment and ability. Companders are sometimes used to correct the problem of inadequate dynamic range transformation. In many situations, the dynamic amplitude range of a signal exceeds the capabilities of its transmission channel, receiver, or restrictions of its environment. These limitations make it desirable to compress the dynamic amplitude range of the signal to allow all portions of the signal to be discerned.
Shown in FIG. 1A is a representative embodiment of a prior art compander, which receives an input signal in both a power estimator circuit and gain multiplier. The typical prior art power estimator provides a linear output to a gain calculate circuit. A control signal may also be provided to the power estimator circuit, typically to modify the attack or release characteristics of the estimator, and the gain calculate circuit, typically to change the amount of compression or expansion. The output of the gain calculate circuit is then combined in the gain multiplier with the input signal to provide a companded signal, taken as an output. A typical power estimator is implemented as a peak detector, for example as shown in FIG. 1B. In FIG. 1B, the input signal is provided to a diode. The output of the diode is tied to ground through an RC circuit, with the output taken at the node connecting the diode, resistor and capacitor. The capacitor charges up to the peak input voltage level and is gradually reduced over time by the resistor. Alternatively, an integrator circuit, also know as a low pass filter, may be used as a power estimator, as shown in FIG. 1C, where the input signal is provided to a conventional RC integrator. Peak detectors and low pass filters suffer from increasing low frequency distortion as the signal frequency approaches the corner frequency of the circuits. Lowering the corner frequency to decrease the low frequency distortion increases the transient response time leading to overamplification and signal clipping and underamplification. Dividing the total bandwidth into multiple frequency bands and using multiple companders can reduce distortion and transient response time but at the cost of the extra processing for the additional companders. A typical prior art stereo compander is shown in FIG. 1D, with left and right input channels each supplied to a multiplier and multiplier value, typically a value of ½. The multiplier outputs are then added and supplied to a pair of conventional, prior art companders. The multipliers and adder form an input signal mixer and is used to maintain relative spatial information in the two channels. The right and left channel input signals are also supplied to the respective companders, with the output of the respective companders being available as the output signal. The input signal mixing typically results in additional distortion since the adjacent channel signal is partly controlling the compander gain. It also typically results in one channel being under-amplified and the other channel being over-amplified which can result in clipping distortion.
Broadcast or recording restrictions or other technological limitations often mandate that the maximum amplitude range of a signal be restricted, and as a result signals are compressed to remain within those restrictions and limitations. After such a signal has been received or recovered, it is often desirable to expand its amplitude range to restore the original dynamic amplitude range. Thus a compressor and expander pair can be used to cancel out low frequency distortion, but do not and cannot function as a standalone compressor or expander.
Referring to FIG. 1E, a prior art compander pair may be better appreciated, where an input signal is provided to a first compander for purposes of compressing the input signal. The compressed signal, which includes some distortion, is then recorded on a suitable media, or otherwise managed. The compressed signal, with distortion, is thereafter provided a second compander in the pair. The second compander expands the compressed signal, and removes the distortion, resulting in restoration of the input signal, which is then provided as the system output signal.
One common limitation of prior art audio signal processing systems is that it is often difficult for a listener to comfortably hear all portions of an audio signal due to environmental audio noise. In such circumstances, when the amplification is sufficient that the louder portions can be discerned easily, the quiet (or low amplitude) portions of the signal are masked by the environmental noise. Environmental noise is transient and often unpredictable in nature, which makes manual adjustment to compensate for it particularly difficult. If the user manually increases the volume for times when the signal cannot be heard due to loud transient noise, the volume will be too loud once when the noise has subsided or the audio signal becomes greater in amplitude.
This problem occurs in many situations, usually (though not always) involving outdoors or mobile environments such as a car stereo, a cellular telephone used in public places, a portable radio used during a public sporting event, or home theater systems. There are also devices such as alarms, door bells, and phone ringers, that cannot be heard at times when there is much environmental noise, or must be set uncomfortably loud in order that the listener is assured of hearing them.
There have been many attempts at solutions to the problem of inadequate noise compensation, a representative embodiment of a prior art solution being shown in FIG. 2.-The environmental noise signal is calculated as the difference between the environmental input (noise plus speaker output) detected by a microphone and a representation of the signal output by the speaker (a.k.a. reference). This environmental noise signal is then provided to a power estimator, typically an integrator or lowpass filter to smooth the signal. The power estimator output is provided to a gain calculate circuit to calculate a gain value to increase the output level of the input signal, typically in a linear manner. This type of prior art solutions provides no means of calibration of the circuitry to the acoustic environment making their proper operation unpredictable. They also do not satisfactorily address problems with variations in the audio source such as long silent pauses in the audio signal, uncontrolled positive feedback known as gain chase, room acoustic resonances that appear as false noise, or allow changing the minimum signal to noise ratio of the system. Further, they have an inadequate response to noise in that some respond too quickly, reacting to phone ringers and short bursts of speech, while others have too long and inaccurate a response. Prior art solutions also do not allow the user to select the signal or noise priority so that if the noise is speech, the signal source will be reduced instead of increased.
Further, when sound is to be heard in multiple locations, each location has a different environmental requirement. The signal necessary to provide adequate sound in one location, often results in sounds that are either too loud or quiet in other areas. This problem is typically solved by the use of redundant equipment, for example one prior art audio system per room.
It is also desirable to have methods to allow a user to perform a number of signal adjustments, e.g. calibration, changing compression or expansion factors, changing the minimum signal to noise ratio, channel balancing and equalization, in order to obtain optimal sound for a given environment. It is even more desirable to have these adjustments done automatically since most users lack the knowledge and understanding of how to perform correct and optimal adjustments to their equipment. Prior art solutions have no automatic adjustments and require manual adjustments with little or no explanation. Several techniques have been developed to address some of these problems.    Blackmer U.S. Pat. No. 3,789,143 (1974) discloses a compander that performs a logarithmic transformation of a signal, proportioned to the root mean square of two 90-degree phase separated signals. While this technique will work for any particular frequency, it is not able to maintain a 90-degree separation over all frequencies.    Beard U.S. Pat. No. 4,169,219 (1979) teaches the use of an analog, low-pass-filter delay-buffer in implementing a compressor-expansion pair, for recording a compressed signal that subsequently would be expanded upon being played. It is unsuitable for standalone use due to significant distortion of low frequency signals.    Bethards U.S. Pat. No. 4,216,427 (1980) instructs in the use of an adaptive audio compressor using analog techniques for restricting subsequent RF signal modulation. Changes in gain occur continuously, resulting in significant distortion of low frequency signals.    Orban U.S. Pat. No. 4,249,042 (1981) shows the use of a multi-frequency band compressor that controls the gain in each band by means of measuring the power in a master band. It has no mechanism to avoid amplifying the noise floor, and no dynamic input signal mapping.    Schroder U.S. Pat. No. 4,306,202 (1981) discloses a discrete compressor and expander, implemented using analog techniques. The compressor and expander modes, selected by analog switches, are intended to be used only in combination with each other. The compressor cannot be used in standalone operation because of distortion of low frequency signals.    Bloy U.S. Pat. No. 4,368,435 (1983) shows the use of a narrow band compander with a combination of a fast and slow attack circuit. By dividing the input into multiple narrow frequency bands, they attempt to minimize signal distortion. This technique provides unacceptable distortion of wide band audio signals.    Unagami et al. U.S. Pat. No. 4,482,973 et al. (1984) is not a compander per se, but instead teaches the use of two AGC (automatic gain control) circuits. Its algorithm does permit its implementation in a DSP and it performs a signal limiting function. The fixed delay causes both synchronization and low frequency response problems.    Stikvoort U.S. Pat. No. 4,562,591 (1985) discloses the use of a peak detector with a non-linear amplifier to provide a compander. Its use of a low-pass filter results in high distortion at low frequencies, and its overall design causes undesirable signal clipping.    Rosback U.S. Pat. No. 4,641,361 (1987) instructs in the use of an analog, multi-frequency band, automatic gain circuit that makes use of a peak clipper to reduce overall gain. Gain is changed continuously, resulting in distortion at low frequencies.    Bloy et al. U.S. Pat. No. 4,853,963 (1989) shows the use of a DSP to process narrow band signals. Its algorithm causes high levels of distortion on wide band signals.    Jorgensen U.S. Pat. No. 4,859,964 (1989) teaches the continual upward and downward adjustment of an automatic gain control to keep an input signal within certain limits. This technique undesirably amplifies the signal noise floor. It does not disclose the algorithms used in the microprocessor.    Thomas U.S. Pat. No. 4,947,133 (1990) instructs in the use of a compressor that uses a fixed signal delay line. After a signal zero crossing occurs, the compressor performs various signal smoothing, gain, sample and hold and compression functions. It performs no signal expansion and is implemented using a combination of analog and digital circuitry.    Akagiri et al. U.S. Pat. No. 4,972,164 (1990) teaches the specific design of a curvilinear compander. It is a complex design that uses both analog and digital circuitry. Its curvilinear algorithm minimizes distortions and abrupt transitions of the companded signal over its entire input range.    Orban U.S. Pat. No. 5,444,788 (1995) shows the use of an analog compander. Its use of diodes in a non-linear low-pass filter causes temperature stability problems and distorted low frequency response.    Werrbach U.S. Pat. No. 5,463,695 (1995) instructs in the use of analog tracking filters to implement a compressor. It performs non-linear compression that compresses transient peaks more than average signals. The average compression is fixed and it does not perform any dynamic range mapping.    Frey et al. U.S. Pat. No. 5,631,968 (1997) discloses an analog design with a variable compression ratio controlled by the time-averaged audio signal and various breakpoints. Low, selected and high compression ratios are used depending on the time-averaged signal. Changes in gain are made continuously, resulting in inherent signal distortion.    U.S. Pat. No. 4,322,579 to Kleis et al. (1982) discloses detecting the environmental noise level. It starts compressing the signal when a certain threshold is reached. The use of a band-pass and high-pass filter results in a substantial reduction in signal fidelity.    U.S. Pat. No. 4,553,257 to Mori et al. (1985) describes an open loop automatic volume control device. It is a single channel, analog circuit that does not alter the signal if the noise is below a given threshold. It performs some amount of variable compression as a function of signal and environmental noise above a given threshold. The primary disadvantage of this invention is that it suffers from positive feedback between the speaker and microphone, making it only useful for headphone applications. It provides no limit on the maximum volume produced.    U.S. Pat. No. 4,628,526 to Germer (1986) teaches using the rate of change of ambient noise and signal, to determine how the signal should be adjusted. If the noise is increasing faster than the signal, it increases the signal amplification, if the signal is increasing faster than the noise, it decreases the signal amplification. It has the desirable feature of not needing user calibration. It has the disadvantage that it always amplifies the signal, when compressing the signal would provide superior comfort to the user. It also requires additional circuitry to address the problem of dealing with silent portions of the audio signal.    U.S. Pat. No. 4,868,881 to Zwicker et al. (1989) shows using a microphone to detect noise. A multi-band equalizer is used to process the noise and audio signals, and the resultant composite signal is amplified and fed back into the noise compensator circuit. The difficulty with this patent is that it requires that the microphone only detect noise, and thus places it in a vehicle engine compartment, when in fact, wind noise is often the dominant source of environmental noise, resulting in the circuit not solving the stated problem.    U.S. Pat. No. 4,882,762 to Waldhaner (1989) teaches the use of a programmable multi-band compression system for hearing aids. It provides different amounts of fixed compression for multiple frequency bands, compensating for hearing loss that is both audio amplitude and frequency dependent. This patent addresses the issue of compressing a signal to compensate for hearing loss. It does not provide variable, automatic compensation of the signal in the presence of environmental noise, since it has no way to detect or distinguish this noise.    U.S. Pat. No. 4,891,837 to Walker et al. (1990) discloses the compression or expansion of a signal for use in a speakerphone. The amount of signal transformation is a function of the ambient noise. The primary source of noise addressed by this invention is the signal received by the microphone from the speakerphone's speaker during a duplex conversation. The invention assumes that the user will speak louder in a room with high ambient noise, and compensates accordingly.    In U.S. Pat. No. 4,953,221 to Holly et al. (1990) shows how positive feedback problems can be avoided by converting noise and audio signals to DC levels and subtracting them from each other. The disadvantage of this technique is that a sample and hold circuit must be used to avoid a noise problem whenever the audio input signal goes silent.    U.S. Pat. No. 5,107,539 to Kato et al. (1992) discloses the means for adjusting the surround or effect sound in a vehicle as environmental noise is sensed. It amplifies, as a function of the sensed noise, the signal and the effect transformation of the signal, using unique level control circuits for each signal. This circuit has several disadvantages, long silent pauses in the audio signal will result in undesirably loud amplification, and it does no compression of the signal, resulting in situations where the audio signal becomes too loud.    U.S. Pat. No. 5,172,358 to Kimura (1992) shows the usage of a digital signal processor to boost low and high frequencies depending on the average amount of sound pressure. No means are provided to calibrate the actual sound pressure with the levels inside the circuit. The controller is the means for controlling the device, yet no algorithms are disclosed for implementing it. There is no mechanism disclosed for handling signals that rapidly increase or decrease.    U.S. Pat. Nos. 5,434,922 and 5,615,270 to Miller et al. (1995/1997) teaches the use of adaptive algorithms with a digital signal processor to determine the amount of noise and dynamically compensate for it. Adaptive and least means square algorithms are computationally intensive and in certain situations can add undesirable amounts of distortion to a signal. The instruction is unclear as how signal processing is performed; FIG. 9 of this patent implies that the invention only performs fixed, 2:1 compression; how to set the minimum limit in item 62; or how to set the compression parameters in the gain calculator shown in item 60. Attack and release are fixed. Usage is made of prior art buffers to perform filter delay compensation. No provision is made to squelch inherent signal source noise. Compression occurs even when there is no environmental noise.    U.S. Pat. No. 5,450,494 to Okubo et al. (1995) shows the use of adaptive filters with a digital signal processor to determine the amount of noise and dynamically compensate for it. Their invention makes use of a fast Fourier transform which is computationally expensive, to determine the coefficients for the adaptive filters. It assumes that noise has a fixed frequency spectrum, dominant in lower frequencies and attenuated at higher frequencies. It teaches the theory of sound pressure and noise.    U.S. Pat. No. 5,509,081 to Kuusama (1996) teaches the use of selective amplification of various frequency bands in order to mask unwanted noise. No signal companding is performed, resulting in circumstances where certain noise-dominant frequency bands are amplified painfully loud. The main distinctive feature is the use of a delay line that is ineffective due to room reverberation and variations caused by changing microphone position. This method works for fixed delays, but does not function well in situations where there is considerable phase delay, signal dispersion or echoes. It provides no instruction on how to calibrate the circuitry.    U.S. Pat. No. 5,530,761 to d'Alayer de Costemore d'Arc (1996) teaches the use of a mathematical algorithm for automatically adjusting sound volume. It performs no signal companding and does not appear to have addressed considerations such as avoidance of gain chase and calibration.    U.S. Pat. No. 5,550,922 to Becker (1996) discloses the use of an analog compressor. It attempts to avoid a gain chase problem by matching the output signal to exceed the environmental noise by a small margin. It provides the means to reduce the gain during signals below a particular threshold. It provides no information on how to measure noise, set the volume or perform calibration.    U.S. Pat. No. 5,666,426 to Helms (1997) discloses the use of a digital signal processing algorithm to provide automatic volume control by maintaining a constant signal to noise ratio. The system calibrates itself by sensing the ambient sound level shortly after being powered on. No signal companding is performed, resulting in circumstances where the output volume is unacceptably loud. It provides instruction on a volume control and calibration, but requires the room to be quiet when calibrating. It does not address normal mode resonances due to room acoustics providing incorrect calibration.    U.S. Pat. No. 4,558,460 to Tanaka, et al. (1985) describes a motor vehicle speed sensor used to increase the output of an amplifier. This will not work in a non-automotive setting since the noise compensation is dependent on vehicle speed and not environmental noise.
As will be appreciated from the foregoing discussion, the prior art companders suffer from a number of disadvantages. All produce high levels of signal distortion at low frequencies. Using multiple frequency band compander techniques to reduce distortion requires significant additional processing requirements. Many rely upon a compressor and expander pair to cancel out low frequency distortion, and do not and cannot function as a standalone compressor or expander. Many of the companders are implemented using analog designs that do not address considerations necessary for the use of digital signal processors, or take advantage of their capabilities. Some use multiple channel compander designs with input signal mixing that result in inter-channel modulation distortion and output clipping. Most compander designs use fixed attack and release times or use a plurality of fixed attack and release time-constant filters, that can result in additional signal distortion or limited operating range. Further, prior art noise compensators suffer from gain chase problems, inadequate and inaccurate response to noise, a lack of signal or noise priority choice, and no means to vary the output signal to noise ratio. Prior art systems typically require redundant equipment and do not provide automatic adjustments.
In summary, existing inventions are ineffective due to inherent design limitations.